vicidial.org

I installed Vicidial. Now, I have some problems. My script call is:
- customers establish a inbound calling to xxx. (xxx is a public exteion of my office)
- They'll hear instruction, after press 2, this call is moved to In-Groups:Sales .If agents in this group aren't available. They'll hear instruction (busy_agent file) and press 1 to be recalled later ( equivalence: Wait Time Option:PRESS_CID_CALLBACK, this is config in INBOUND -> show in-group ->Sales)

Problems are:
+Case A: When Vicidial is connected SIP trunk to VOIP_GATEWAY, and VOIP_GATEWAY has FXO port to connect PSTN. My scenario is executed successfuly. This log is:

Executing [8@Maint:1] AGI in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Started music on hold, class 'default', on SIP/trunk1
Parsing '/etc/asterisk/manager.conf': Found
Manager 'sendcron' logged on from 127.0.0.1
Parsing '/etc/asterisk/manager.conf': Found
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
-- Stopped music on hold on SIP/trunk1
-- Started music on hold, class 'default', on SIP/trunk1
Parsing '/etc/asterisk/manager.conf': Found
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
-- Stopped music on hold on SIP/trunk1
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=1) (sample_offset 0)
-- Playing 'busy_agent' (escape_digits=1) (sample_offset 0)
[Mar 18 09:32:16] DTMF14178: channel.c:2530 __ast_read: DTMF begin '1' received on SIP/trunk1
[Mar 18 09:32:16] DTMF14178: channel.c:2534 __ast_read: DTMF begin ignored '1' on SIP/trunk1
[Mar 18 09:32:16] DTMF14178: channel.c:2449 __ast_read: DTMF end '1' received on SIP/trunk1, duration 250 ms
[Mar 18 09:32:16] DTMF14178: channel.c:2515 __ast_read: DTMF end passthrough '1' on SIP/trunk1
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'vm-goodbye' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0

+Case B: When Vicidial is connected SIP trunk to ISP. customers call to xxx,and press 2, establish successfully.But agents aren't available. They cant't hear busy_agent file, just hear music on hold until timout. This is log:

Executing [8@Maint:1] AGI in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=1) (sample_offset 0)
-- Playing 'busy_agent' (escape_digits=1) (sample_offset 0)
-- Started music on hold, class 'default', on SIP/sip-trunkISP
Parsing '/etc/asterisk/manager.conf': Found
Manager 'sendcron' logged on from 127.0.0.1
Parsing '/etc/asterisk/manager.conf': Found
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
Parsing '/etc/asterisk/manager.conf': Found
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
-- Stopped music on hold on SIP/sip-trunkISP
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=1) (sample_offset 0)
-- Playing 'busy_agent' (escape_digits=1) (sample_offset 0)
-- Started music on hold, class 'default', on SIP/sip-trunkISP == Refreshing DNS lookups.
-- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0

As you see, with only inbound-group Sales, but two scripts are different completely. Case A is ok. But Case B, script plays busy_agent two times. But I can't hear this sound
I wonder what diffrent bewteen two connections : Vicidial_ISP and Vicidial_VOIPGateway_COLine
Please help me. Thanks in advance.

I do not see an "agent busy" recording being played in the first example. Thus when there is no "agent busy" in the second example, it is not "different". Both merely appear to play MOH.

You can attempt to use "agi debug" in the asterisk cli to find differences.

The "agent busy" recording you mention to "busy_agent" file
In this log's asterisk:
" Playing 'busy_agent' (escape_digits=1) (sample_offset 0)"

Questions:

Are you sure that is your only difference? You JUST changed the voip carrier from a gateway voip to an external SIP carrer? Same ingroup, phone number calling, everything? Just that one change?

Next: Explain for me what "8@Maint:1" represents (all pieces of it!). I assume you have a call menu named Maint ... but if the customers are pressing 2, what is the 8?

I also note that it does show the same "busy_agent" in both the good and bad examples. So I'm wondering if it is a timing issue where there is a delay on the gateway call that causes the busy agent recording to occur after the call has moved to another stage somehow. But first it would be good to verify that this is an accurate CLI output when there was no audio. Since the audio appears in the CLI and does not show a "file not found" result ... usually this means the sound did get sent. Of course, there could also be some unusual codec transcoding issue to the parallel audio path.

And then there's my standard response, which I neglected earlier but will resolve now

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

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